similar to: SIP, NAT and Firewalls

Thanks Moises. I was kind of hoping that, at least if I hosted my Asterisk server somewhere where there was no NAT for the * box that the SIP phones wouldn't create any issues. How do you people with Hosted PBX handle the deployment of SIP phones behind NAT firewalls? Is it just elbow grease and configuring every single phone for the customer, or is there a way? Mike you can redirect

Mark, Thanks a lot for the feedback. It's reassuring to say the least Mike Message: 18 Date: Sat, 21 Jan 2006 15:36:18 -0500 From: Mark Phillips <g7ltt@g7ltt.com> Subject: Re: [Asterisk-Users] SIP and NAT - best practices? To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <43D29B42.3060705@g7ltt.com> Content-Type:

Hi, I'm looking to implement Fax reception on a SIP line. I`ve been looking at the Wiki and some other web pages and it`s far from clear what I need to do, or if it`s even possible. 1) Is it possible, or does it only work on Zap channels? (as I`ve read somewhere) 2) Is there a good reference on the web to do so? Thanks, Michael

Hi, I've asked this question in the past, but I didn't get a precise answer. Hopefully somebody will take note of my question. Before I forget, I am using Asterisk 1.2.4. I've been using the Voicemail app with success (i.e. it works) except for one single thing: the ONLY message that it ever played back to the caller is the temporary message. If I delete the temporary message

Hi, I've had a bit of a problem with one way audio, and it happens exactly when I believe it shouldn't (and works perfectly when I would guess I could have issues. Setup: GrandStream GXP2000-------Linksys Router-----------Internet------Asterisk box (hosted somewhere, fixed IP, no NAT) ----------- VoIP provider -------PSTN When a call comes in from the PSTN, the call goes all the way

You will probably have to build that yourself, or really customize something off the shelf. Depending on what phones you are using you might be able to do that via the phones xml interface. Have fun with that I would be interested to see how it goes. -- Justin -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of

Yes I did. Fair question. I think it`s working, but is there anyway to know for sure? Show modules show app_cdr.so as existing... Mike On Thursday 26 Jan 2006 16:50, Micha?l Gaudette wrote: > Hi, > > I've just reinstalled Asterisk 1.2.3 on a fresh system and since I've > noticed that the CDR logging in MySQL (on a different computer) has > stopped. I thought it

Is it not for a card with 4 FXO? I spent several hours the other day trying to figure out what I had done wrong and I ahd forgotten to connect the power cable. I setup several of these before and couldn't figure out why this one didn't work. It appears that's all it waqs. Without the power connecter the card will probe, and even appear to be working but when the lines ring (coming

We are doing this with the latest spandsp, iaxmodem and hylafax. Seems to work very well for us so far. -Jonathan > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Michael Gaudette > Sent: Tuesday, March 21, 2006 3:34 PM > To: 'Asterisk Users Mailing List - Non-Commercial

I don't know if this has been mentioned yet: Voicepulse is now offering wholesale pricing and IAX2 connectivity for Asterisk users. No fees, pay as you go. They also offer incoming calls for $7.99 per month. See wholesale.voicepulse.com.

Hello, a local prison contacted us regarding some calling card solution. they need 4 E1s to serve 120 rooms in that prison. we are planning on using 4 servers to serve the calls and one for the database servers' specifications are: 2.8 Dual Core Proccessors 2 GB Ram 160 Sata Drive each server will be provided with 1 E1 card Questions are: 1- will those servers be able to handle that ammount

That was my thought too. Even if it *does* work without it, there may be a reason internally you can't see why it is required. (Ex. Putting extra stress on a component causing it to fail in 6 months) -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tom Rymes Sent: Wednesday, August 10, 2005 3:49 PM To:

Hey everybody, I have a situation where we have 2 Asterisk (CVS as of 08/25/2005) connected via IAX. On the corporate side, we have 1 TE110P connecting to a Definity G3R and it's connecting to a TN464F card, giving a 23 channel connection. I have echocancel=yes, echotraining=yes and echocancelwhenbridged=yes. One the remote office side, they a Adit 600 channel bank for 10 outside

Hi, I`ve recently upgraded a server from 1.8 to Asterisk 13. While everything is under control, I have one issue with the way CDRs are kept for queues. And I don`t mean ?I don`t like it?. I mean it crashes the server. I realize there are multiple CDRs per queue call ? one per ring/per phone, basically. The issue is that whenever the number of CDRs ?to be recorded? for a call exceeds 5000,

I recently did a set up where I replaced a simple D-link home router that was having trouble processing a T1's worth of bandwidth with a linux machine running iptables. the kernel was 2.6.29-r5 and I chose the SIP connection tracking modules from the menuconfig. Router worked fine for normal traffic, but I was unable to get the SIP phones to work. Using ngrep it was plain to see

Hello In case Asterisk is used in a private LAN behind a firewall while allowing remote SIP clients to connect from the Net, we must open UDP5060 for SIP and a range of UDP ports (as set in rtp.conf) so let incoming voice packets. Provided the user doesn't have access to the firewall (eg. corporate or hotel), and the firewall doesn't allow dynamic port opening through UPnP or NAT-PMP...

I've started to implement checks for package versions on dependencies in install.packages(). However, this is revealing a number of problems/misconceptions. (A) We do not check versions when loading namespaces, ahd the namespace registry does not contain version information. So that for example (rtracklayer) Depends: R (>= 2.7.0), Biobase, methods, RCurl Imports: XML (>=

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Anyone here using Level 3 or AT&T wholesale sip terminations services? I would like to know on any minimums they would require? Also, an idea of how competitive the rates are. I am not asking to disclose your custom rate deck, just a "what to expect". Finally, if you guys can PM me contact info to someone from the wholesale department, I would really appreciate it. Kind Regards,

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